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• ffmpeg error : Too large number of skipped frames xxxxx > 60000

  16 septembre 2019, par Fulkron

  I try to record on RasperryPI from usb Cam (/dev/video0) using segments to obtain sliced video.
  The command I’m using is :

  ffmpeg -input_format mjpeg -video_size 640x480 -i /dev/video0 -y -y -f v4l2 -an -pix_fmt yuv420p -codec copy -f segment -segment_time 60 -segment_wrap 10 -t 180 stream/looper%02d.avi

  I got an error after 2 minutes about skipped frames (see later for detail).
  Is I run the same command without segmentation is working.

  I tested on different HW, with different segment parameters, with different usb cam but the situation still the same.

  Here there is the output :

     ffmpeg version 4.1.4-1+rpt1~deb10u1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8 (Raspbian 8.3.0-6+rpi1)
  
    configuration: --prefix=/usr --extra-version='1+rpt1~deb10u1' --toolchain=hardened --libdir=/usr/lib/arm-linux-gnueabihf --incdir=/usr/include/arm-linux-gnueabihf --arch=arm --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-omx-rpi --enable-mmal --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  Input #0, video4linux2,v4l2, from '/dev/video0':
  
    Duration: N/A, start: 40185.241073, bitrate: N/A
  
      Stream #0:0: Video: mjpeg, yuvj422p(pc, bt470bg/unknown/unknown), 640x480, 25 fps, 25 tbr, 1000k tbn, 1000k tbc
  
  [segment @ 0x9800b0] Opening 'stream/looper00.avi' for writing
  
  Output #0, segment, to 'stream/looper%02d.avi':
  
    Metadata:
  
      encoder         : Lavf58.20.100
  
      Stream #0:0: Video: mjpeg, yuvj422p(pc, bt470bg/unknown/unknown), 640x480, q=2-31, 25 fps, 25 tbr, 600 tbn, 1000k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [segment @ 0x9800b0] Opening 'stream/looper01.avi' for writing/A speed=1.01x    
  
  [segment @ 0x9800b0] Opening 'stream/looper02.avi' for writing/A speed=1.01x    
  
  [avi @ 0x9822a0] Too large number of skipped frames 72034 > 60000speed=1.01x    
  
  av_interleaved_write_frame(): Invalid argument
  
  [avi @ 0x9822a0] Too large number of skipped frames 72034 > 60000
  
  frame= 3011 fps= 25 q=-1.0 Lsize=N/A time=00:02:00.06 bitrate=N/A speed=1.01x 

  I did not find any suggestion to solve it.
  Could you help me ?

  Thanks
  Dario

• FFMPEG APC to MP3 Command

  7 octobre 2019, par Kuja

  Files compressed at a rate of 32 KBit / s using a special adpcm codec. How do I convert this file to mp3 ?

  ffmpeg -i "10_____________AO905326704670____10_________.apc"   audio.mp3
  
  
  
  ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  
    configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  10_____________AO905326704670____10_________.apc: Invalid data found when processing input

• Invalid data found when processing file with ffmpeg API

  18 septembre 2019, par Timmy K

  I have an m4a file containing two AAC-encoded audio streams. I want to decode it on Android with the ffmpeg library. However, avformat_open_input() fails with "Invalid data found when processing input".

  This file was created (i.e. the audio data encoded) with the same library.

  I’ve double-checked the accuracy of the file path I provide to the call.

  The file doesn’t appear to be corrupt. It plays with QuickTime player, and ffprobe reports the output below.

  I’ve looked at the source in ffmpeg’s utils.c but it’s not clear at which of several points (including nested calls) this invalidity of the data in the file is being decided.

  FFPROBE OUTPUT
  
  built with gcc 7.2.0 (crosstool-NG fa8859cb)
  
  configuration: --prefix=/home/ubuntu/miniconda3 --disable-doc --enable- 
  
  shared --extra-cflags='-fPIC -I/home/ubuntu/miniconda3/include' --extra-
  
  cxxflags='=-fPIC' --extra-libs='-L/home/ubuntu/miniconda3/lib -lz' --enable-
  
  pic --disable-static --disable-gpl --disable-nonfree --disable-openssl --
  
  enable-libvpx --cc=/opt/conda/conda- 
  
  bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-cc --cxx=/opt/conda/conda-bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-c++ --enable-libopus
  
      libavutil            55. 78.100 / 55. 78.100
  
      libavcodec         57.107.100 / 57.107.100
  
      libavformat        57. 83.100 / 57. 83.100
  
      libavdevice        57. 10.100 / 57. 10.100
  
      libavfilter         6.107.100 /    6.107.100
  
      libswscale            4.    8.100 /    4.    8.100
  
      libswresample     2.    9.100 /    2.    9.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'recording-2019-09-03-110626.m4a':
  
      Metadata:
  
          major_brand         : isom
  
          minor_version     : 512
  
          compatible_brands: isomiso2mp41
  
          encoder                 : Lavf58.27.103
  
      Duration: 00:00:05.04, start: 0.000000, bitrate: 637 kb/s
  
          Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 314 kb/s (default)
  
          Metadata:
  
              handler_name        : SoundHandler
  
          Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 315 kb/s (default)
  
          Metadata:
  
              handler_name        : SoundHandler

  The call that fails with "invalid data found" is :

   if ( (ret = avformat_open_input( &inputFormatContext, filePath, 0, 0)) < 0 )
  
    {
  
      __android_log_print(ANDROID_LOG_DEBUG, "MyTag", "Could not open input file '%s' error %s ", filePath, av_err2str(ret) );
  
        return false;
  
    }

  I would expect this to "just work" since I encoded and muxed the two audio tracks with the same build of the ffmpeg libraries.

  When I built the libraries, I configured the build with :

  —enable-decoder=aac,pcm_s16le —enable-encoder=aac,pcm_s16le —enable-demuxer=mp4,wav —enable-muxer=mp4,wav —enable-protocol=file,http

  When I iterate through the codecs using av_codec_iterate() and look for supported sample rates I see :

  Found codec aac
  
  supported sample rate 96000
  
  supported sample rate 88200
  
  supported sample rate 64000
  
  supported sample rate 48000
  
  supported sample rate 44100
  
  supported sample rate 32000
  
  supported sample rate 24000
  
  supported sample rate 22050
  
  supported sample rate 16000
  
  supported sample rate 12000
  
  supported sample rate 11025
  
  supported sample rate 8000
  
  supported sample rate 7350
  
  Found codec pcm_s16le
  
  Found codec aac
  
  Found codec pcm_s16le 

  Each codec appears twice, which I thought was maybe the encoder and decoder. But I note that only one has any supported sample rates. My test files’ sample rate is 44100.

  Any thoughts on what other things I can check for ? I wondered whether I should create and pass an AVInputFormat object explicitly to the call, based upon what I know about the contents of the file. But I don’t know how to construct one.